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#include <libda/plugins/audio_dev.hpp>
#include <gst/gst.h>
#include <ostream>
#include <stdint.h>

#include <iostream>

namespace {
      using namespace da;
      class gst_record: public record::dev {
            static void c_callback(GstElement*, GstBuffer* buffer, GstPad*, gpointer userdata) {
                  gst_record& self = *static_cast<gst_record*>(userdata);
                  int16_t const* iptr = reinterpret_cast<int16_t const*>(GST_BUFFER_DATA(buffer));
                  try {
                        // Warning GST_BUFFER_SIZE returns a size in byte and not in frames
                        std::vector<sample_t> buf(GST_BUFFER_SIZE(buffer)/sizeof(int16_t));
                        std::transform(iptr, iptr + buf.size(), buf.begin(), conv_from_s16);
                        std::size_t channels = self.s.channels();
                        pcm_data data(&buf[0], (GST_BUFFER_SIZE(buffer) /sizeof(int16_t)) / channels, channels);
                        self.s.callback()(data, self.s);
                  } catch (std::exception& e) {
                        self.s.debug(std::string("Exception from recording callback: ") + e.what());
            settings s;
            struct init {
                  init() {
                        GError* e = NULL;
                        if (!gst_init_check(NULL, NULL, &e)) {
                              std::string msg = std::string("GStreamer could not be initialized: ") + e->message;
                              throw std::runtime_error(msg);
                  // gst cannot be deinitialized safely :(   ~init() { gst_deinit(); }
            } initialize;
            GstElement* pipeline;
            gst_record(settings& s_): s(s_), initialize() {
                  // FIXME: this code probably has cleanup trouble in case of exceptions

                  pipeline = gst_pipeline_new("record-pipeline");

                  GstElement* source;
                  if (!(source = gst_element_factory_make("alsasrc", "record-source")))
                    if (!(source = gst_element_factory_make("osssrc", "record-source")))
                    if (!(source = gst_element_factory_make("osxaudiosrc", "record-source")))
                    throw std::runtime_error("Cannot create record source");

                  GstElement* audioconvert;
                  if (!(audioconvert = gst_element_factory_make("audioconvert", NULL)))
                    throw std::runtime_error("Cannot create audioconvert");

                  GstElement* audioresample;
                  if (!(audioresample = gst_element_factory_make("audioresample", NULL)))
                    throw std::runtime_error("Cannot create audioresample");

                  GstElement* sink;
                  if (!(sink = gst_element_factory_make("fakesink", "record-sink")))
                    throw std::runtime_error("Cannot create fakesink");
                  gst_bin_add_many(GST_BIN(pipeline), source, audioconvert, audioresample, sink, NULL);
                  g_object_set(G_OBJECT(sink), "sync", TRUE, NULL);
                  g_object_set(G_OBJECT(sink), "signal-handoffs", TRUE, NULL);
                  g_signal_connect(G_OBJECT(sink), "handoff", G_CALLBACK(c_callback), this);
                  /* Link the elements together */
                  GstCaps* caps = gst_caps_new_simple(
                    "rate", G_TYPE_INT, s.rate(),
                    "width", G_TYPE_INT, 16,
                    "depth", G_TYPE_INT, 16,
                    "channels", G_TYPE_INT, s.channels(), NULL);
                  if (!gst_element_link_many(source, audioconvert, audioresample, NULL))
                    throw std::runtime_error("Cannot link the GStreamer elements together ('src' -> 'audioconvert' -> 'audioresample')");

                  if (!gst_element_link_filtered(audioresample, sink, caps))
                    throw std::runtime_error("Cannot link the GStreamer elements together ('audioresample' -> 'fakesink')");

                  /* TODO: gst_element_set_state(_pipeline, GST_STATE_PAUSED); */
                  gst_element_set_state(pipeline, GST_STATE_PLAYING);
            ~gst_record() {
                  if (!pipeline) return;
                  gst_element_set_state(pipeline, GST_STATE_NULL);
      plugin::simple<record_plugin, gst_record> r(devinfo("gst", "GStreamer PCM capture. Settings are not used."));

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